similar to: Transcoding CPU usage: surveys?

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

For about 10 months I have been running a succession of Asterisk SVN trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2 at my home. I have a variety of SIP phones (mostly Polycom) internally; my external connections are two POTS lines on a TDM400P (with HPEC) and an IAX2 link to a VoIP provider. I had Asterisk configured to allow G.722 and u-law on the Polycom phones,

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as

Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig

Sirs/Ladies, Not sure if anyone saw anything like that before... I was playing with an Asterisk setup with a Grandstream BT101 (1.0.5.11) and www.voipjet.com (IAX2). The other devices I have home (Sipura 3k and DTA-310) seem to work just fine, but the Grandstream seems to suffer from one-way voice (remote end can't hear me). The only workaround I found so far (have not spoken with VOIPJet

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my